The present invention relates to a data network and to a method of communicating data over the network. Particularly, the invention relates to real time data over Internet Protocol such as Voice over Internet Protocol (VoIP) networks.
The architectural model of Internet telephony is rather different to that of the traditional telephone network. All signaling and media flows run over a “best effort” Internet Protocol (IP) based network (either the public Internet or various private intranets), and in appearance any device can communicate directly with any other.
Internet telephony can currently be used for free (or almost) between two (or more) Real Time Data over IP “hosts” (e.g., PC's gateways, etc.) using public IP-based networks (the Internet, e.g., through a dial up connection or not). However, the quality of the audio/video is highly dependent on the actual loading of the network.
In order to yield good quality, the audio/video media flows preferably has a short end-to-end latency. Such real time flows do not co-habit very well on the same IP-based network with various flows used for the transfer of data. Hence, ideally real time flows need to run on separate IP-based networks with little or a carefully controlled amount of data traffic.
The operator of such a separate Real Time Data over IP network will usually demand a compensation for the use of his infrastructure. The subscriber to the service may have to pay a fee for the duration, the distance and the type of session. Hence, the Real Time Data over IP network must accurately determine the status of each session, even in case of a failing host terminal or other abnormal situations, e.g., attempted fraud.
A Real Time Data over IP host can be any native IP device (e.g., H.323/SIP terminal), multimedia PC or ordinary telephone which sets up a session/call over the IP-network with another Real Time Data over IP host in order to make, e.g., a telephone call or a combination of a telephone call with data, i.e., a multimedia session.
A distinction can be made between an ordinary telephone, a H.323/SIP terminal (e.g., a native IP terminal), and a multimedia device (i.e., PC), as each may have a different type of connection to the network and support different types of services.
The telephone and multimedia devices are usually directly connected to a telephone exchange. These devices therefore usually have an indirect connection to the IP-network. A telephone, used for IP-telephony, is connected to a Media Gateway of its Internet Telephony Service provider (ITSP VoIP provider) via the telephone exchange. On the other hand, a multimedia PC interconnects via the telephone exchange and a remote access service (RAS) function of an internet service provider (ISP) to the network of the ITSP, usually using a dial-up (modem) connection. The possibility therefore exists to make a VoIP call, using suitable software, e.g., Netmeeting, Net2Phone, etc.
Native VoIP terminals are directly connected to the VoIP network, so that, for example, the H.323/SIP terminals support the encoding/decoding and packetization/sequencing of information (i.e. the Gateway Function) exchanged with other H.323 terminals.
In the prior art, a media gateway is used to connect two dissimilar networks, for example, a Public Switched Telephony Network (PSTN) and an IP network. This connectivity of dissimilar networks is achieved by translating protocols for call set-up and release, converting media formats between the different networks, and transferring information between the networks connected by the gateway.
In the prior art, a call server is used which is a device located in the data network, controlling and handling the voice over data streams from PSTN and data users to other PSTN and data users. The transport functions of the call server include interfaces to the data network for the transport of the voice streams. Signaling protocols interface with both the PSTN and data network.
Typical Real Time Data over IP connection procedures at present are described here below for some Voice over IP (VoIP) examples:
(1) Using a telephone: A telephone user dials into the Media Gateway of a VoIP provider. As explained earlier, voice will be transformed into VoIP packets and sent over the IP-network to a destination VoIP host.
(2) Using a multimedia PC: A PC dials into the RAS of an ISP provider. Via the ISP, the PC-user is able to set up a VoIP connection. This is possible with a direct connection to another VoIP host or via a VoIP provider.
(3) Using a native IP device: a VoIP host using protocols such as H.323 or SIP, initiates a session by requesting this to a Call Server which has a Gatekeeper function. The Gatekeeper maintains a list of subscribers and grants (or not) the permission to proceed—e.g., after the verification of the identity or the account of the subscriber. It also returns the current IP address of the destination. In the case that the called party is within the PSTN, the session will be directed to a PSTN/IP gateway, the latter playing the role of the terminating VoIP host. The VoIP hosts will then negotiate session set-up and capabilities with each other using elements of the above protocol. Since the gatekeeper needs to be informed at all times about the status of the session, the VoIP hosts initially route all signaling through the gatekeeper (the gatekeeper in effect proxies these messages). Once the call set up is complete, the media flows run directly between the VoIP hosts.
Though this architecture behaves well and the gatekeeper is usually able to make accurate call duration records (CDR's) in normal cases, it is not able to defend itself against failing terminals, attempted fraud (spoofing a call release without actually doing so) or even abusing the Real Time Data over IP network for other traffic not related to the Real Time Data over IP.